[Asterisk-Users] SIP rtp port forcing

[boris.vincent at mindspeed.com]

When a SIP call starts (INVITE / 200 OK), asterisk seems to create a 
random port number for voice (rtp) packets. Is it possible to force this 
port value (without using reinvite since i am trying to use SIP against 
something else than sip)

thanks a lot in advance
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